The present invention generally relates to communications networks and, more particularly, relates to high bandwidth communications networks, including Voice-over-Internet Protocol (VoIP) communications networks, and bandwidth loading and capacity limiting systems and methods therefor.
Voice-over-Internet Protocol (VoIP) telephony communications networks typically require significant bandwidth for operations, particularly where numerous VoIP calls are being concurrently made over the networks. Quality of VoIP service, as well as access to the service, is highly dependent on sufficient network bandwidth capacity. Conventionally, VoIP communications services have been limited because of the bandwidth requirements for such services, and also because of network congestion and usages by all network communications occurring concurrent with the services.
Network bandwidth has conventionally been available for network communications on a relatively ad hoc basis—that is, network communications use up whatever bandwidth may be available, from time to time, for the communications. Because VoIP communications often require high bandwidth capacities for quality and access of such services, there have been no assurances that sufficient bandwidth may, at any point, be available for suitable VoIP communications over the networks and branches of the networks. As with automobile traffic on roadways, bottlenecks and traffic slow-downs and congestion occur when sufficient bandwidth (e.g., roadway) is not available and traffic is attempting to flow. This has been the case with network bandwidth in network communications, and particularly applies where VoIP communications (and pluralities of such communications) are occurring concurrent with all other communications signal traffic over the network.
Packetized communications networks, such as, for example, the Internet in accordance with Transport Communications Protocol/Internet Protocol (TCP/IP) protocols or other networks according to other applicable network protocols, experience congestion of traffic where the traffic is not regulated or restricted. Typically, there have not been any particular means or elements to effect capacity limitations over communications networks, where the networks are employed simultaneously for VoIP communications and other uses. Providers of the network communications have been unable to guarantee or assure availability of particular bandwidth on networks as may be required by particular network communicators. Particularly where VoIP communications are mission critical services, providers have not been able to assure availability and quality of the VoIP communications because all are subject to network state and available capacities from time to time.
In VoIP communications network systems, analog voice signals are digitized and transmitted as a stream of packets over a digital data network. These systems enable real-time transmission of the voice signals as packetized data over networks that employ digital network communications protocols, including Transport Control Protocol (TCP), Real-Time Transport Protocol (RTP), User Datagram Protocol (UDP), and other Internet Protocol (IP) and network protocol suites. The digital networks that carry VoIP include the Internet and other digital data communications channels, such as public and private wired or wireless networks, WAN, LAN, WLAN, Wi-Fi, intranets/extranets, enterprise networks, and other conventional communications connectors and implementations and combinations thereof.
In the IP networks for voice communications, individual digital data packets are constructed corresponding to analog voice signals. The packets follow independent paths over the networks to the intended destination. Packets associated with a particular source in such networks can, thus, take many different paths to the destination for the packets. The packets can arrive at the destination with different delays, arrive out of sequence, or not arrive at all. The destination for the packets must re-assemble and convert the packets back to original analog voice signals.
The current VoIP communications can comply with several possible standards, and it is expected that varied and additional standards will arise. The most commonly followed standard at present appears to be the ITU-T H.323 standard, although other standards employed include H.248, H.323, IETF, ITU, IETF TFC 2885, Media Gateway Control Protocol (MGCP), and Session Initiation Protocol or IETF RFC 2543 (SIP), among others. Generally, each of the various standards in VoIP implementations do not encompass all aspects of VoIP communications. The variations among the various standards, and also the networks and equipment therewith employed, include algorithms, bandwidth limitations, packet loss recovery, compression, speech processing, and other concepts for improved communications, efficiencies, and speed.
In general, the VoIP technology (whatever it may be, in the particular circumstance) allows voice calls originated and terminated on standard telephones supported by the public switched telephone network (PSTN) and IP communications endpoints (e.g. IP phone) to be conveyed over IP networks. Gateways for the VoIP digital data packets provide the bridge between the local PSTN and the IP network, at both the originating and terminating sides of a VoIP call. In the case of a call that is originated and terminated on the PSTN, to originate a call, the calling party accesses a nearby gateway, either by a direct connection or by placing a common analog call over the local PSTN and entering the desired destination phone number. The VoIP technology translates the destination telephone number into a network address, i.e., an IP address, associated with a specific terminating gateway at the destination of the call. At the terminating gateway, a call is initiated to the destination phone number over the local PSTN to establish end-to-end two-way communications. Thereafter, the analog voice signals entered on each end are digitized into packets and communicated over the packet network at each respective transmitting gateway, and the digitized packets so communicated are reassembled and translated back into the analog voice signals corresponding to the received packets from the network at each respective terminating gateway. In the case where one or multiple of the end-points is not the PSTN an IP communications endpoint (e.g., communications gateway) replaces the gateway in the call flow, and performs the same function of converting voice (or other communications methods) to IP format data signals.
Because quality and availability of VoIP communications over data networks is highly dependent on adequacy, quality and sufficiency of network capacity, it would be a significant advantage and improvement in the art and technology to provide systems and methods for assuring available network capacity for use in making VoIP telephony calls over the network. The present invention provides capacity limiting features to communications network operations, in order to make better available network bandwidth for VoIP communications or other mission critical bandwidth usages. Furthermore, the present invention provides numerous advantages and improvements in the art and technology, including by making quality and availability decisions based on higher-level applications and not only the lower-level network itself.